DEAL Free U.S. DIDs Unlimtd Channels

kdthomas

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I got it working guys. It was the NAT config that I needed to add. This is now easily added under Settings/SIP Setting/NAT Setting (General/Chan SIP Settings). What I noticed is there is no ring. When I dial from my cell to the new conference line it picks up but I never hear a ring. Not complaining...just never experienced that when I dial through my Google trunks.
 

Jose Pinto

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This Forum is powerfull and we can see this in the DID offer.
Today they change the page, and now has options that we can choose about services that we maybe yes or maybe not intersted and also they just has this: "I understand that if ordered numbers have less than 1 minute usage within 60 days, they will be assigned to another client automatically.*"

Nice
 

kdthomas

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They contacted me today to up sale me on outbound calling. Unlimited inbound is free, but outbound they charge for. That is fine by me. I just needed a free way to do unlimited inbound for my conference line since Google restricts you to 2-3 channels. If you don't plan to use yours very often just set a reminder (ala "Hey Siri remind me every 60 days"). :)
 

Jose Pinto

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Hi @kdthomas
Yes we can do this but I have other thing that can be done, I have 4 emails that is at Google Gmail and I setup the 2 time passwords one is a phone call so I can give this number so they will call me everytime that I will need to check my email, this is another option.
 
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wardmundy

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This Forum is powerfull and we can see this in the DID offer.
Today they change the page, and now has options that we can choose about services that we maybe yes or maybe not intersted and also they just has this: "I understand that if ordered numbers have less than 1 minute usage within 60 days, they will be assigned to another client automatically.*"

Nice
That's what Telephone Reminders are for. :cool: Send yourself a call to each of your DIDs once a month using the recurring reminder feature.
 

dicko

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Actually you will need to call each DID and have it ANSWERED for at least 61 seconds every month. The reminders won't do that .
 

wardmundy

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Actually you can send any length message you want with our Reminders app. 20 seconds every week would do fine. Thanks for catching the 60 second requirement though.

Better yet, here is a message that you can send once a month to a destination that will automatically answer or route to voicemail. It's over one minute long so you'll never have to worry:

"This is a simple message to preserve our phone number: 800-555-1212. Have a nice day. This is a simple message to preserve our phone number: 800-555-1212. Have a nice day. This is a simple message to preserve our phone number: 800-555-1212. Have a nice day."
 
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restamp

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It seems I'm always late to these parties. I ordered a couple DIDs over 24 hours ago but have not heard anything yet. However, I notice that the offer site now reads
Code:
Order will be rejected if the provided IP is a local IP, make sure you provide a valid static IP that is accessible publicly.
Does anyone know what I need to do to make my site "accessible publicly" (src IP, ports)? I don't want to turn off iptables.
 

wardmundy

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@restamp Just follow the steps in the first post. Your IP needs to be a public IP and, with Incredible PBX, you need to WhiteList the IP address of the provider's server for SIP: UDP 5060 using add-ip.
 

restamp

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Actually, my concern was that TXC was going to try to ping my server, or do something similar to prove there really was a PBX at the address I had given them, and would cancel my request when my box ignored them. However, I got my "Welcome Packet" later in the day today, so it turns out my concerns were unwarranted.

That said, I honestly could not get the DIDs to work using the conventional chan_SIP trunking page in the GUI. I believe the reason is that TXC always sends an INVITE request to port *5060* when setting up an incoming call and port 5060 is owned and operated by pjsip on my system (Incredible PBX 13-12.3). To work around this problem, I resorted to creating the TXC trunk under pjsip, something I'd never done before. Pjsip doesn't have a "Peer Details" window like the conventional SIP and the correlations weren't obvious. The important setting is "SIP Server", which should be set to 216.66.23.179 (or whatever address TXC gives you).

This got the DIDs working, but it isn't smooth: Each time Asterisk reloads, pjsip attempts to register with TXC. This attempt is rejected, and semi-permanently blocked if "Permanent Auth Rejection" is checked. The other glitch is with Qualifies: The URI that pjsip cobbles together for the OPTIONS request ('sip:mad:216.66.23.179:5060') is invalid. I bumped the qualify interval up to a large value to minimize the error messages that pjsip was dumping into the logs every minute. But the important thing is that both my DIDs are now working.

If anyone can explain how to clean up the Qualifies under pjsip, or better yet how to convert this trunk to the more conventional Chan_SIP, I'm all ears, but for now it's working well enough to fiddle around with.
 

dicko

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There is no reason to try to "register" hrere, the connection is inbound only and sent on 5060 to the IP you told them to send it to, It will not replay to anything you try to send it, so don't try, it will confuse you and be totally ignored by them.
 
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restamp

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Unfortunately, dicko, there is one compelling reason for my Asterisk to attempt this registration -- namely I have yet to figure out how to turn it off (at least until it has tried to register once). Any ideas? Remember, this is a beta chan_pjsip I'm using, not the chan_sip we all know and love.

FWIW, I just tried a small conference call and the new DID easily handled four simultaneous sessions. Nice of TXC to offer them to us.
 

Jose Pinto

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Hi all
Nice surprise just now (saturday, 15:00 GNT -2).
As I decide to work with phones again ( I already did this in past from1989 till 2009) I just order DIDs from IPC LLC and I order Iowa number, Texas, Vermont and California and I only did not receive the Vermont number.
Today I was testing California number since I receive it in email they say that the number has no 1 first, So you need to write an inbound route without the the number 1, because if you don't do that it does not work. Anyway I was trying to call it ( I just forget to took the 1, so it does not work till I change the route) and as it does not work I was preparing myself to change the route when I receive a call - funny!!! an USA call? I was thinking who is calling me as I did not advertising the numbers, for my surprise it was a call from IPC LLC guy - Brian Shepard - this is a familiar name to all that already made a DID order, when we receive the DID his name and email comes with it.
He just call me to tell me about the need of absense of the number 1 in my Inbound Route.
I was very surprise with his kindness in call me to assist me. So I would like to write here to all you guys know how the company works.
My special thanks to Brian and to all guys here in this board for all help and support. Thank you very much.

Ps: If anyone here read my first post will see that I start in octuber with Incredible and also using C&C, is very easy to see all dificult that I had. But now all my extensions is working, my routes, my Voips and my new DIDs,an believe or not I have not a single problem, and still in C&C, of course I also have others Servers (that I saw here) and I only have problems with Woothosting. For all informations that I already got here: Thank you very much.
 
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dicko

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Unfortunately, dicko, there is one compelling reason for my Asterisk to attempt this registration -- namely I have yet to figure out how to turn it off (at least until it has tried to register once). Any ideas? Remember, this is a beta chan_pjsip I'm using, not the chan_sip we all know and love.

FWIW, I just tried a small conference call and the new DID easily handled four simultaneous sessions. Nice of TXC to offer them to us.
Just don't turn it on :) your trunk should have no "outbound" settings.
 
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Jose Pinto

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Hi all
I need a little help (please remember I'm in a learnning process).
To receive SMS with this DID is it necessary to ask to add TXT only?
How do I configure Enchilada to receive SMS with this DID?
Can someone here help me?
Thank you for your time and attention
 
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mardosas

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Unfortunately, I cannot get the numbers working. Below are the Asterisk logs. Unnecessary to say that fail2ban is blocking the ip during the call.


Trunk configuration made according to Nerd’s article. Did I miss something?


2018-01-31 07:38:41] NOTICE[16495]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '"1347-mycallerid" <sip:[email protected] serve rip>' failed for '216.66.23.179:5060' (callid: [email protected]:5060) - No matching endpoint found

[2018-01-31 07:38:41] NOTICE[16495]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '"1347-mycallerid" <sip:[email protected]>' failed for '216.66.23.179:5060' (callid: [email protected]:5060) - Failed to authenticate
 

tycho

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@mardosas All I had to do was this:

(1) I used the "add IP" script to whitelist the provider's IP (accomplishes what @Jose Pinto is talking about above)
(2) I added a trunk that I called telcomsxchange with the following in "Peer Details":

type=friend
qualify=yes
host=216.66.23.179
context=from-trunk​

(3) I created an inbound route for my DID(s) pointing to my extension of choice.

Works fine.
 
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mardosas

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This is actually strange and no help yet. IP whitelisted, otherwise there would be no reaction from Asterisk. Peer details exactly as above. Still not working.
 

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